[Freeswitch-users] garbled audio with G726-32, other codecs are fine

Michael Jerris mike at jerris.com
Mon Aug 12 18:14:45 MSD 2013


There is always bit packing, but there are 2 different ways to do the bit packing.  A lot of devices get it wrong so its worth looking at that.

Mike

On Aug 4, 2013, at 1:53 AM, Ivan Mitev <imitev at c3i.bg> wrote:

> Thanks for the suggestion but I'm testing with G726-32, not AAL2-G726-32 
> ; so bitpacking shouldn't be used. By the way when I tested with AAL2 to 
> the linphone client I only got cracks and whitenoise, I've forgot to 
> mention that in my post.
> 
> That said I've tried to uncomment and set <param name="bitpacking" 
> value="none"/> in internal.xml ("none" is a wild guess - I couldn't find 
> any doc on values accepted by this parameter), but that doesn't help.
> 
> Speex: the ATAs don't support it. And being stubborn I'd like to 
> understand what's wrong with G726 :)
> 
> 
> On 08/04/2013 05:22 AM, Jeff Leung wrote:
>> 
>> You can turn off G726 AAC bit-packing in spandsp.conf.xml.
>> 
>> By the way, there are other codecs out there you can try. SPEEX comes 
>> to mind if all your endpoints don’t deal with the PSTN.
>> 
>> *From:*freeswitch-users-bounces at lists.freeswitch.org 
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of 
>> *Brian Foster
>> *Sent:* Saturday, August 3, 2013 2:37 PM
>> *To:* FreeSWITCH Users Help
>> *Subject:* Re: [Freeswitch-users] garbled audio with G726-32, other 
>> codecs are fine
>> 
>> AAC bitpacking by any chance? I thought I had a similar issue, 
>> happened so long ago I cant remember what I did.
>> 
>> Thank you,
>> 
>> Brian Foster
>> Project Manager/Owner's Rep.
>> Davri Investments, Inc.
>> O: 317-787-2686 x2102
>> M: 317-600-9753
>> E: bdfoster at davri.com <mailto:bdfoster at davri.com>
>> Indianapolis, Indiana
>> 
>> Sent from a mobile device.
>> 
>> On Aug 3, 2013 5:20 PM, "Ivan Mitev" <imitev at c3i.bg 
>> <mailto:imitev at c3i.bg>> wrote:
>> 
>> Hello
>> 
>> I'm migrating an office setup from asterisk to FS and in the process I
>> was considering using G726-32 for some bandwidth starved remote
>> endpoints. However I only get metallic/garbled audio with that codec
>> even when simply playing moh to the endpoint, while other codecs work
>> fine (G711U/A, G722, GSM). G732-16 is inaudible, G732-40 sounds
>> marginally better but still garbled and really worse than G711.
>> 
>> The setup is FS 1.2.12 from FS' yum repo on a centos6 64bit KVM guest
>> (centos6 64bit host). But please don't shoot ! :) - I know about virtual
>> environment limitations but for these tests the host is only lightly
>> loaded, there aren't any calls to the FS instance except my tests, and
>> the fact that it works with other codecs makes me think that
>> virtualization is not the issue here. I may be wrong though.
>> 
>> Is there any guide for debugging that kind of problem before reverting
>> to a fresh install on bare-metal with the latest HEAD ? Until now I've
>> tried:
>> 
>> - improving timers ; but the default soft timer (which I guess uses
>> timerd) works best. The time interval between sent packets on a tcpdump
>> trace looks identical to the output of "timer_test", so that doesn't
>> seem to be a network/jitter problem. And there's no problem with other
>> codecs, but maybe G726-XX is specific. For info the guest's clocksource
>> is kvm_clock, while the host uses tsc.
>> 
>> - using different endpoints: the production ones are Linksys PAP2
>> ("fixed" for 20ms psize, and G726-32 SDP type indentification), but the
>> same thing happens with linphone on a fedora 19 laptop.
>> 
>> A call with rtp media going through FS without transcoding - G726-32 to
>> G726-32 - works perfectly (I can't hear the difference with G711). The
>> problem is only when there's transcoding to G726 (from wav for moh, or
>> from any other codec when bridging). I've looked at the wiki, posts,
>> changelogs, jira, ..., but am a bit at a loss now.
>> 
>> Any pointers ?
>> 
>> Except that little problem, FS rocks, and I'm happy I can finally ditch
>> asterisk. Kudos to the core devs and contributors.
>> 
>> Ivan
>> 




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